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gerrit-public.fairphone.software / platform / external / webrtc / c58c8a54224d4e2269b69c6218d2a0d89a2f650f
  1. 0f54f21 Removes deprecated GetSentPacket from PacketResult. by Sebastian Jansson · 6 years ago
  2. dc98b9b AEC3: Corrected include by Per Åhgren · 6 years ago
  3. c564a7b Roll chromium_revision 7841106b37..793c8566ab (605607:605715) by chromium-webrtc-autoroll · 6 years ago
  4. 8ffd710 Update Android encoder to use GetEncoderInfo() by Erik Språng · 6 years ago
  5. 020e583 AEC3: Compensate comfort noise level for loss due to filter bank by Gustaf Ullberg · 6 years ago
  6. 83b00f0 AEC3: Computation of comfort noise gains from suppression gains by Gustaf Ullberg · 6 years ago
  7. 34fc346 Add support for computing iOS code coverage by Artem Titarenko · 6 years ago
  8. 277b6ea Isolating APM API build target: adding dummy :api target. by Alessio Bazzica · 6 years ago
  9. 3ddaf3c Revert "Add support for screen sharing with PipeWire on Wayland" by Patrik Höglund · 6 years ago
  10. 82c07ea Tune huge video frames detection threshold for GetStats googHugeFramesSent stat by Ilya Nikolaevskiy · 6 years ago
  11. 4f3cc6e Make VideoSendStreamTest.NoPaddingWhenVideoIsMuted less flaky by Erik Språng · 6 years ago
  12. a8f5461 nit: Use make_unique in rtp_video_stream_receiver.cc by Elad Alon · 6 years ago
  13. 27f3172 Simplify use of events in TestAudioDevice by Niels Möller · 6 years ago
  14. 361dbc1 Android: Add option to set presentation timestamp in EglRenderer by Magnus Jedvert · 6 years ago
  15. 967f7d5 Add audio level to CSRC class by Jonas Oreland · 6 years ago
  16. df351f4 Update FakeEncoder to use EncoderInfo by Erik Språng · 6 years ago
  17. 254d3db Add missing #include to absl/memory/memory.h from audio_encoder_cng.cc by tzik · 6 years ago
  18. fbf1683 Add HdrMetadata to VideoFrame by Johannes Kron · 6 years ago
  19. 4f0f3d5 Remove unused member variable - RTCPSender::using_nack_ by Elad Alon · 6 years ago
  20. 63ada78 Remove outdated TODO by Sam Zackrisson · 6 years ago
  21. 3ea1878 Add severity into RTC logging callbacks by Jiawei Ou · 6 years ago
  22. edfb883 Roll chromium_revision 11d7305a72..7841106b37 (605505:605607) by chromium-webrtc-autoroll · 6 years ago
  23. d7db17b Roll chromium_revision bf7ad46dee..11d7305a72 (605401:605505) by chromium-webrtc-autoroll · 6 years ago
  24. a9bbd86 Add a configuration parameter for using the media transport for data channels. by Bjorn Mellem · 6 years ago
  25. 41b5296 Roll chromium_revision c26ff44a53..bf7ad46dee (605286:605401) by chromium-webrtc-autoroll · 6 years ago
  26. ee49f70 Remove VideoEncoder::SetChannelParameters. by philipel · 6 years ago
  27. c22f551 Remove locks from AECM and move it into private_submodules_ by Sam Zackrisson · 6 years ago
  28. e693381 Delete struct rtc::PacketTime. by Niels Möller · 6 years ago
  29. 0070655 Removing ancient and unused test scripts and data files by Henrik Lundin · 6 years ago
  30. fd1a2fb Set RtpRtcp config receive_only in voe::ChannelReceive by Niels Möller · 6 years ago
  31. aed3070 Replace GetScalingSettings() with GetEnocderInfo() in TestEncoder by Erik Språng · 6 years ago
  32. f418bcb Refactor RtpSender to use absl::string_view for payload name. by Niels Möller · 6 years ago
  33. 2634199 Move MovingAverage to rtc_base/numerics and update it. by Ilya Nikolaevskiy · 6 years ago
  34. a1ead6f Update EncoderProxy to use EncoderInfo by Erik Språng · 6 years ago
  35. bf0d0c1 Add IPv6 configuration parameters to iOS API by Uladzislau Susha · 6 years ago
  36. 842a2a8 Roll chromium_revision 4e7c87b55c..c26ff44a53 (605184:605286) by chromium-webrtc-autoroll · 6 years ago
  37. e7547d5 Move MemoryStream to separate source files, and to a test target. by Niels Möller · 6 years ago
  38. 9f878f6 Roll chromium_revision b58a03341b..4e7c87b55c (605082:605184) by chromium-webrtc-autoroll · 6 years ago
  39. 671341a Roll chromium_revision 35f882550d..b58a03341b (604980:605082) by chromium-webrtc-autoroll · 6 years ago
  40. 1bc0b9d Roll chromium_revision e842ab5f98..35f882550d (604874:604980) by chromium-webrtc-autoroll · 6 years ago
  41. 2039ee7 Revert "Delete rtc::Pathname" by Qingsi Wang · 6 years ago
  42. 273d029 Implement data channel methods in LoopbackMediaTransport. by Bjorn Mellem · 6 years ago
  43. 0367d1a Adds a field trial parameter to configure waiting time before sending Nack packets. by Ying Wang · 6 years ago
  44. e401863 Change to RtcEvent::Copy by Elad Alon · 6 years ago
  45. 2365936 Hide the AudioEncoderCng class behind a create function by Karl Wiberg · 6 years ago
  46. 42e7d9c Enable rtc event log in *_loopback tools running with renderers by Ilya Nikolaevskiy · 6 years ago
  47. f8ba95e Add field trial for vp8 cpu speed configuration for arm. by Åsa Persson · 6 years ago
  48. 56ef305 Move event logging of config into AudioSendStream. by Oskar Sundbom · 6 years ago
  49. 6bf2054 Roll chromium_revision 734e273d43..e842ab5f98 (604373:604874) by chromium-webrtc-autoroll · 6 years ago
  50. aa3c1cc Delete _strnicmp. Uses replaced with abseil functions. by Niels Möller · 6 years ago
  51. 41f00de Fix chromium roll by Artem Titov · 6 years ago
  52. 6b9dec0 Delete rtc::Pathname by Niels Möller · 6 years ago
  53. d4a68bd Implement Injectable Audio Codecs for the Java SDK. by Lennart Kolmodin · 6 years ago
  54. 3e4c77f Fix AGC2 fixed-adaptive gain controllers order. by Alessio Bazzica · 6 years ago
  55. 096d016 Update MultiplexEncoderAdapter to use EncoderInfo by Erik Språng · 6 years ago
  56. 58df0ad Add guards to VideoCaptureDS::Init for when pins are null by Andreas Pehrson · 6 years ago
  57. 9b5b070 Use EncoderInfo in SimulcastEncoderAdapter by Erik Språng · 6 years ago
  58. 4eb4112 Plug-in media transport state listener by Piotr (Peter) Slatala · 6 years ago
  59. 189013b Update QualityTestVideoEncoder to use GetEncoderInfo() by Erik Språng · 6 years ago
  60. 449afd9 Updated ScopedVideoEncoder to use GetEncoderInfo() by Erik Språng · 6 years ago
  61. 5e78461 Make the extra seturation margin configurable. by Alex Loiko · 6 years ago
  62. b1e031a JitterEstimator: Remove old LowRate exp and add trial for upper bound. by Erik Språng · 6 years ago
  63. 15ca5a9 Add implicit conversion between rtc:PacketTime and int64_t. by Niels Möller · 6 years ago
  64. 96965ae Add ability to enable frame dumping decoder via field trial. by Erik Språng · 6 years ago
  65. fe45da4 Remove WebRTC-VP8-GfBoost field trial. by philipel · 6 years ago
  66. af6d741 Makes send time information in feedback non-optional. by Sebastian Jansson · 6 years ago
  67. be837ac Remove RTPFragmentationHeader from LibvpxVp8Encoder. by philipel · 6 years ago
  68. 2812763 Remove deprecated AudioProcessing::GetStatistics function by Sam Zackrisson · 6 years ago
  69. 4e93329 Properly setup MockPeerConnectionObserver in tests (continued). by Yves Gerey · 6 years ago
  70. dd20c9c Add support for screen sharing with PipeWire on Wayland by Tomas Popela · 6 years ago
  71. 7f4dfa4 Remove locks from AEC2 and move it into private_submodules_ by Sam Zackrisson · 6 years ago
  72. 59844ce Revert "Use the factory instead of using the builtin code path in `VideoCodecInitializer`." by Qingsi Wang · 6 years ago
  73. 7852d29 Improve the documentation of MdnsResponderInterface and rename MDns.* to Mdns.*. by Qingsi Wang · 6 years ago
  74. eb2c641 Delete the default implementations of MediaTransportInterface methods. by Bjorn Mellem · 6 years ago
  75. be14217 Use the factory instead of using the builtin code path in `VideoCodecInitializer`. by Jiawei Ou · 6 years ago
  76. 8386435 Roll chromium_revision 6271fcdc14..734e273d43 (604273:604373) by chromium-webrtc-autoroll · 6 years ago
  77. 1f6aa9f Add interfaces for using MediaTransport as the transport for data channels. by Bjorn Mellem · 6 years ago
  78. 062a691 Roll chromium_revision 9996ac8918..6271fcdc14 (604166:604273) by chromium-webrtc-autoroll · 6 years ago
  79. 9f95625 When SDES is used, pass pre-shared key to media transport. by Piotr (Peter) Slatala · 6 years ago
  80. 7182286 Allow FakeNetworkPipe to wake up its processing thread by Sebastian Jansson · 6 years ago
  81. 693432d Add obj-c mapping from native configuration to RTCConfiguration by Piotr (Peter) Slatala · 6 years ago
  82. e6caa9f export RTCRtpTransceiverInit by Piasy · 6 years ago
  83. ed45c57 Corrects audio overhead correction in Scenario test. by Sebastian Jansson · 6 years ago
  84. 69807e8 Depend directly on destination targets. by Yves Gerey · 6 years ago
  85. a8fa2d0 Move some methods from StreamInterface to FifoBuffer by Niels Möller · 6 years ago
  86. 21cddff Harmonize paths to dependent targets. by Yves Gerey · 6 years ago
  87. b32bb95 Bugfix: FlexFEC causes retransmit bitrate increase. by Ying Wang · 6 years ago
  88. 8b7d206 AEC3: Decrease latency until the delay has been detected by Per Åhgren · 6 years ago
  89. f577ab3 Roll chromium_revision 7e85c0922c..9996ac8918 (604065:604166) by chromium-webrtc-autoroll · 6 years ago
  90. b00b28e Roll chromium_revision 0cb3899c4e..7e85c0922c (603959:604065) by chromium-webrtc-autoroll · 6 years ago
  91. b3f887b Expose key derivation through a simple interface for use in WebRTC. by Benjamin Wright · 6 years ago
  92. 1a92cd7 Roll chromium_revision 34bb9a9162..0cb3899c4e (603839:603959) by chromium-webrtc-autoroll · 6 years ago
  93. c78b0ea Create a MediaTransportState enum and add a state callback to MediaTransport. by Bjorn Mellem · 6 years ago
  94. eaf337a Remove event wait logic from DesktopConfigurationMonitor by Emircan Uysaler · 6 years ago
  95. 746d46b AGC2: renaming GainCurveApplier to Limiter. by Alessio Bazzica · 6 years ago
  96. fcc3981 Revert "Use only first payload timestamp for RTCP SR generation for audio" by Ilya Nikolaevskiy · 6 years ago
  97. 992a868 Fix for clock reset repair. by Christoffer Rodbro · 6 years ago
  98. a2e133d Delete StreamInterface::ReadLine. by Niels Möller · 6 years ago
  99. ed7b8b1 Update media transport settings struct by Piotr (Peter) Slatala · 6 years ago
  100. 3e67676 Add support for field trials in peerconnection_client|server by Bjorn Terelius · 6 years ago